[svn-commits] may: branch may/ooh323_ipv6_direct_rtp r349927 - in /team/may/ooh323_ipv6_dir...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Jan 6 15:13:17 CST 2012
Author: may
Date: Fri Jan 6 15:13:13 2012
New Revision: 349927
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=349927
Log:
fix blobs
Modified:
team/may/ooh323_ipv6_direct_rtp/addons/ooh323c/src/ooh245.c
team/may/ooh323_ipv6_direct_rtp/configs/ooh323.conf.sample
Modified: team/may/ooh323_ipv6_direct_rtp/addons/ooh323c/src/ooh245.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_ipv6_direct_rtp/addons/ooh323c/src/ooh245.c?view=diff&rev=349927&r1=349926&r2=349927
==============================================================================
--- team/may/ooh323_ipv6_direct_rtp/addons/ooh323c/src/ooh245.c (original)
+++ team/may/ooh323_ipv6_direct_rtp/addons/ooh323c/src/ooh245.c Fri Jan 6 15:13:13 2012
@@ -3264,7 +3264,7 @@
OOTRACEDBGC3("Empty TCS found. (%s, %s)\n",
call->callType, call->callToken);
- ooH245AcknowledgeTerminalCapabilitySet(call);
+ ooH245AcknowledgeTerminalCapabilitySet(call);
call->remoteTermCapSeqNo = tcs->sequenceNumber;
/* close all transmit chans */
Modified: team/may/ooh323_ipv6_direct_rtp/configs/ooh323.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_ipv6_direct_rtp/configs/ooh323.conf.sample?view=diff&rev=349927&r1=349926&r2=349927
==============================================================================
--- team/may/ooh323_ipv6_direct_rtp/configs/ooh323.conf.sample (original)
+++ team/may/ooh323_ipv6_direct_rtp/configs/ooh323.conf.sample Fri Jan 6 15:13:13 2012
@@ -27,13 +27,13 @@
; OOH323/exten/peer OR OOH323/exten at ip
;
; Domain name resolution is not yet supported.
-;
+;
; When a H.323 user calls into asterisk, his H323ID is matched with the profile
; name and context is determined to route the call
;
-; The channel driver will register all global aliases and aliases defined in
+; The channel driver will register all global aliases and aliases defined in
; peer profiles with the gatekeeper, if one exists. So, that when someone
-; outside our pbx (non-user) calls an extension, gatekeeper will route that
+; outside our pbx (non-user) calls an extension, gatekeeper will route that
; call to our asterisk box, from where it will be routed as per dial plan.
@@ -47,9 +47,9 @@
;The dotted IP address asterisk should listen on for incoming H323
;connections
;Default - tries to find out local ip address on it's own
-bindaddr=0.0.0.0
-
-;This parameter indicates whether channel driver should register with
+bindaddr=0.0.0.0
+
+;This parameter indicates whether channel driver should register with
;gatekeeper as a gateway or an endpoint.
;Default - no
;gateway=no
@@ -65,7 +65,7 @@
;H323-ID to be used for asterisk server
;Default - Asterisk PBX
-h323id=ObjSysAsterisk
+h323id=ObjSysAsterisk
e164=100
;CallerID to use for calls
@@ -128,7 +128,7 @@
; CNG tone or an incoming T.38 RequestMode packet
;
; yes - enable both detection (CNG & T.38)
-; no - disable both
+; no - disable both
; cng - enable CNG detection (default)
; t38 - enable T.38 request detection
;
@@ -137,7 +137,7 @@
; User/peer/friend definitions:
; User config options Peer config options
; ------------------ -------------------
-; context
+; context
; disallow disallow
; allow allow
; accountcode accountcode
@@ -174,7 +174,7 @@
context=context1
disallow=all
allow=gsm
-allow=ulaw
+allow=ulaw
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